a-Reverb: Revert some of previous changes, sound of reverb much improved
authorDamien Zammit <damien@zamaudio.com>
Wed, 13 Jul 2016 09:30:03 +0000 (19:30 +1000)
committerDamien Zammit <damien@zamaudio.com>
Wed, 13 Jul 2016 09:30:03 +0000 (19:30 +1000)
libs/plugins/a-reverb.lv2/a-reverb.c
libs/plugins/a-reverb.lv2/a-reverb.ttl.in

index 944fe22bb2e5c4f6b2fd653d456a61bf1b56f0a1..ef7ace0ae60e8120bafac5b7f5d2a2a559b2387b 100644 (file)
@@ -25,7 +25,7 @@
 #include <math.h>
 #include <string.h>
 
-#define RV_NZ (8+4)
+#define RV_NZ 7
 #define DENORMAL_PROTECT (1e-14)
 
 
@@ -53,7 +53,7 @@ typedef struct {
 static int
 setReverbPointers (b_reverb *r, int i, int c, const double rate)
 {
-       int e = (r->end[c][i] * rate / 44100.0);
+       int e = (r->end[c][i] * rate / 25000.0);
        e = e | 1;
        r->delays[c][i] = (float*)realloc ((void*)r->delays[c][i], (e + 2) * sizeof (float));
        if (!r->delays[c][i]) {
@@ -79,52 +79,37 @@ initReverb (b_reverb *r, const double rate)
        r->dry = 0.7;
 
        /* feedback combfilter */
-       r->gain[0] = 0.75;
-       r->gain[1] = 0.75;
-       r->gain[2] = 0.75;
-       r->gain[3] = 0.75;
-       r->gain[4] = 0.75;
-       r->gain[5] = 0.75;
-       r->gain[6] = 0.75;
-       r->gain[7] = 0.75;
+       r->gain[0] = 0.773;
+       r->gain[1] = 0.802;
+       r->gain[2] = 0.753;
+       r->gain[3] = 0.733;
 
        /* all-pass filter */
-       r->gain[8] = 0.5;
-       r->gain[9] = 0.5;
-       r->gain[10] = 0.5;
-       r->gain[11] = 0.5;
+       r->gain[4] = sqrtf (0.5);
+       r->gain[5] = sqrtf (0.5);
+       r->gain[6] = sqrtf (0.5);
 
        /* delay lines left */
-       r->end[0][0] = 1116;
-       r->end[0][1] = 1188;
-       r->end[0][2] = 1277;
-       r->end[0][3] = 1356;
-       r->end[0][4] = 1422;
-       r->end[0][5] = 1491;
-       r->end[0][6] = 1557;
-       r->end[0][7] = 1617;
+       r->end[0][0] = 1687;
+       r->end[0][1] = 1601;
+       r->end[0][2] = 2053;
+       r->end[0][3] = 2251;
 
        /* all pass filters left */
-       r->end[0][8] = 556;
-       r->end[0][9] = 441;
-       r->end[0][10] = 341;
-       r->end[0][11] = 225;
+       r->end[0][4] = 347;
+       r->end[0][5] = 113;
+       r->end[0][6] = 37;
 
        /* delay lines right */
-       r->end[1][0] = 1116 + stereowidth;
-       r->end[1][1] = 1188 + stereowidth;
-       r->end[1][2] = 1277 + stereowidth;
-       r->end[1][3] = 1356 + stereowidth;
-       r->end[1][4] = 1422 + stereowidth;
-       r->end[1][5] = 1491 + stereowidth;
-       r->end[1][6] = 1557 + stereowidth;
-       r->end[1][7] = 1617 + stereowidth;
-
-       /* all pass filters */
-       r->end[1][8] = 556 + stereowidth;
-       r->end[1][9] = 441 + stereowidth;
-       r->end[1][10] = 341 + stereowidth;
-       r->end[1][11] = 225 + stereowidth;
+       r->end[1][0] = 1687 + stereowidth;
+       r->end[1][1] = 1601 + stereowidth;
+       r->end[1][2] = 2053 + stereowidth;
+       r->end[1][3] = 2251 + stereowidth;
+
+       /* all pass filters right */
+       r->end[0][4] = 347 + stereowidth;
+       r->end[0][5] = 113 + stereowidth;
+       r->end[0][6] = 37 + stereowidth;
 
        for (int i = 0; i < RV_NZ; ++i) {
                r->delays[0][i] = NULL;
@@ -185,7 +170,7 @@ reverb (b_reverb* r,
                /* First we do four feedback comb filters (ie parallel delay lines,
                 * each with a single tap at the end that feeds back at the start) */
 
-               for (j = 0; j < 8; ++j) {
+               for (j = 0; j < 4; ++j) {
                        y = *idxp0[j];
                        *idxp0[j] = x0 + (gain[j] * y);
                        if (endp0[j] <= ++(idxp0[j])) {
@@ -193,7 +178,7 @@ reverb (b_reverb* r,
                        }
                        xa += y;
                }
-               for (; j < 12; ++j) {
+               for (; j < 7; ++j) {
                        y = *idxp0[j];
                        *idxp0[j] = gain[j] * (xa + y);
                        if (endp0[j] <= ++(idxp0[j])) {
@@ -208,7 +193,7 @@ reverb (b_reverb* r,
 
                *yp0++ = ((wet * y) + (dry * xo0));
 
-               for (j = 0; j < 8; ++j) {
+               for (j = 0; j < 4; ++j) {
                        y = *idxp1[j];
                        *idxp1[j] = x1 + (gain[j] * y);
                        if (endp1[j] <= ++(idxp1[j])) {
@@ -216,7 +201,7 @@ reverb (b_reverb* r,
                        }
                        xb += y;
                }
-               for (; j < 12; ++j) {
+               for (; j < 7; ++j) {
                        y = *idxp1[j];
                        *idxp1[j] = gain[j] * (xb + y);
                        if (endp1[j] <= ++(idxp1[j])) {
@@ -337,14 +322,10 @@ run (LV2_Handle instance, uint32_t n_samples)
        }
        if (*self->roomsz != self->v_roomsz) {
                self->v_roomsz = *self->roomsz;
-               self->r.gain[0] = self->v_roomsz;
-               self->r.gain[1] = self->v_roomsz;
-               self->r.gain[2] = self->v_roomsz;
-               self->r.gain[3] = self->v_roomsz;
-               self->r.gain[4] = self->v_roomsz;
-               self->r.gain[5] = self->v_roomsz;
-               self->r.gain[6] = self->v_roomsz;
-               self->r.gain[7] = self->v_roomsz;
+               self->r.gain[0] = 0.773 * self->v_roomsz;
+               self->r.gain[1] = 0.802 * self->v_roomsz;
+               self->r.gain[2] = 0.753 * self->v_roomsz;
+               self->r.gain[3] = 0.733 * self->v_roomsz;
        }
 
        reverb (&self->r, input0, input1, output0, output1, n_samples);
index dabb988c6fd66b4ce7cae4dda2b1dc8275527bec..c8f7252cd0a37060396f64ae9270201d3ed88026 100644 (file)
@@ -70,5 +70,5 @@
                lv2:name "Room Size";
                lv2:default 0.5;
                lv2:minimum 0.5 ;
-               lv2:maximum 0.8 ;
+               lv2:maximum 1.0 ;
        ] .